MPEG-4: a multimedia compression standard for interactiveapplications and services

The Moving Pictures Experts Group (MPEG), which produced the MPEG-1 and MPEG-2 video and audio compression standards, is developing the MPEG-4 standard. MPEG-4 targets interactive multimedia applications and will become a standard in 1999. As well as an increased compression efficiency, MPEG-4 will also offer content-based functionality, i.e. the possibility of accessing and manipulating individual objects in the picture. Furthermore, MPEG-4 will offer possibilities for efficient video storage and for transmission over poor audio and video channels at bit rates between 5 kbit/s and 4 Mbit/s. This paper gives an overview of the state of the art of MPEG-4 development, concentrating especially on video content-based functionality, which is so important for interactive applications     

基于DirectShow的H.264视频采集与播放的实现

H.264标准与现有的视频压缩标准相比有诸多优势,它以优异的压缩性能在数字电视广播、视频实时通信、网络视频流媒体以及多媒体短信各个方面发挥重要作用。ffdshow是windows平台用于DirectShow和VFW进行多媒体播放的非常出色的工具,ffdshow的H.264编码采用的是X264编码器,X264是一个实用的H.264编码器。DirectShow提供对多媒体数据流的高质量捕获和回放,代表着未来多媒体应用程序开发的方向。本文使用DirectShow技术调用ffdshow实现了H.264视频的采集与播放。这种视频编解码方法可用于远程医疗、视频监控和视频会议等多媒体应用。     

Intranet.tv — Automated Multimedia Content Delivery

Intranet.tv is a Web-based platform that allows simple management and delivery of streamed multimedia content, by building on existing Internet protocols. This paper considers the state of the art for streaming media, by introducing the technology behind intranet.tv, and the value-add that intranet.tv provides to corporate video-streaming.     

Internet protocols for multimedia communications. II. Resourcereservation, transport, and application protocols

Part I (ibid., July-Oct. 1997) surveyed the evolution of Internet protocols and applications and described the Internet protocol IPv6 in detail. This part discusses new developments at the upper layers that support real-time Internet multimedia, such as audio and video conferencing and shared whiteboard applications. Application-level framing (ALF), proposed in 1990 for protocol and application design, now forms the basis for many new Internet protocols and applications, including Real-time Transport Protocol (RTP) and Mbone multimedia applications. RTP supports real-time applications that adapt to changing network situations to maintain the quality of service (QoS). The Resource Reservation Protocol (RSVP) provides new Internet services with higher quality than best-effort by means of resource reservations     

Application of H.263+ Video Coding Modes in Lossy Packet Network Environments

The quality of real-time audio and video information transmitted via today's Internet suffers severely from often significant packet losses. While this problem is well understood and solved for existing audio coding schemes, support from the video coding standards themselves is required for video streams. This paper presents the newly introduced error resilience mechanisms built into the second version of H.263 (1998), known under its working name H.263+, and addresses the corresponding packetization format issues that together significantly improve the image quality at packet loss rates up to 20%. In particular, it is support from the video coding algorithm itself, paired with appropriate transport layer mechanisms, that leads to significant improvements of perceived image quality for communicative as well as retrieval applications at moderate bit rates up to some 100 kbit/s.     

基于嵌入式流媒体服务器的视频点播系统

移动通信网络流媒体服务资费较高,限制了人们在没有有线电视、宽带互联网连接、无线WiFi互联网连接的地方用手机收看视音频节目.针对这一现状,提出了一种基于嵌入式流媒体服务器的多功能视频点播系统.以ARM板和WiFi网络适配器作为流媒体服务器,设计一款Android手机播放器作为客户端,实现在智能手机上、多个客户端同时免流量接收服务器本地或外接设备推送的高清、超高清视频.该系统适用于公交、广场、电梯、商场、景区等场合,人们用手机观看视音频,实现不用耗费自己的流量,自由快速地选择自己喜欢的节目.     

A survey on TCP-friendly congestion control

New trends in communication, in particular the deployment of multicast and real-time audio/video streaming applications, are likely to increase the percentage of non-TCP traffic in the Internet. These applications rarely perform congestion control in a TCP-friendly manner; they do not share the available bandwidth fairly with applications built on TCP, such as Web browsers, FTP, or e-mail clients. The Internet community strongly fears that the current evolution could lead to congestion collapse and starvation of TCP traffic. For this reason, TCP-friendly protocols are being developed that behave fairly with respect to coexistent TCP flows. We present a survey of current approaches to TCP friendliness and discuss their characteristics. Both unicast and multicast congestion control protocols are examined, and an evaluation of the different approaches is presented     

Video Streaming with Network Coding

Recent years have witnessed an explosive growth in multimedia streaming applications over the Internet. Notably, Content Delivery Networks (CDN) and Peer-to-Peer (P2P) networks have emerged as two effective paradigms for delivering multimedia contents over the Internet. One salient feature shared between these two networks is the inherent support for path diversity streaming where a receiver receives multiple streams simultaneously on different network paths as a result of having multiple senders. In this paper, we propose a network coding framework for efficient video streaming in CDNs and P2P networks in which, multiple servers/peers are employed to simultaneously stream a video to a single receiver. We show that network coding techniques can (a) eliminate the need for tight synchronization between the senders, (b) be integrated easily with TCP, and (c) reduce server’s storage in CDN settings. Importantly, we propose the Hierarchical Network Coding (HNC) technique to be used with scalable video bit stream to combat bandwidth fluctuation on the Internet. Simulations demonstrate that under certain scenarios, our proposed network coding techniques can result in bandwidth saving up to 60% over the traditional schemes.     

基于Android平台流媒体播放器的研究与实现

为了解决Android自身多媒体库支持的视频文件格式少、编解码效率相对较低的问题,提出一种流媒体播放器的实现方案,并详细阐述了其实现过程。该方案是基于集录制、转换音/视频编解码功能为一体的音视频流整合方案FFmpeg,通过在Linux环境下对其进行裁剪和优化,编译并移植到Android平台上,实现了本地视频以及远程实时视频监控的播放功能。实验结果表明,此Android流媒体播放器能实现流畅的播放,占内存小,可嵌入其它应用程序中,具有广泛的应用前景。     

Cross-layer architecture for scalable video transmission in wireless network

Multimedia applications such as video conference, digital video broadcasting (DVB), and streaming video and audio have been gaining popularity during last years and the trend has been to allocate these services more and more also on mobile users. The demand of quality of service (QoS) for multimedia raises huge challenges on the network design, not only concerning the physical bandwidth but also the protocol design and services. One of the goals for system design is to provide efficient solutions for adaptive multimedia transmission over different access networks in all-IP environment. The joint source and channel coding (JSCC/D) approach has already given promising results in optimizing multimedia transmission. However, in practice, arranging the required control mechanism and delivering the required side information through network and protocol stack have caused problems and quite often the impact of network has been neglected in studies. In this paper we propose efficient cross-layer communication methods and protocol architecture in order to transmit the control information and to optimize the multimedia transmission over wireless and wired IP networks. We also apply this architecture to the more specific case of streaming of scalable video streams. Scalable video coding has been an active research topic recently and it offers simple and flexible solutions for video transmission over heterogeneous networks to heterogeneous terminals. In addition it provides easy adaptation to varying transmission conditions. In this paper we illustrate how scalable video transmission can be improved with efficient use of the proposed cross-layer design, adaptation mechanisms and control information.     

应用于P2P网络串流的实时监控系统

点对点(Peer-to-Peer,P2P)传输架构的新技术能利用用户的网络带宽以及运算能力来分担串流服务系统的负担,对于该网络进行实时监控已经成为近年来的一个研究热点.分析了应用于P2P网络串流的实时监控系统,指出跟踪监控的3类主要方法,即网络级别、被动的应用程序级别和主动应用程序级别的监控.给出了主动应用程序级别的监控模式下监控系统的架构与组成,并通过实验分析了其监控性能.     

Reducing Channel Capacity for Scalable Video Coding in a Distributed Network

In recent years, the development of multimedia devices has meant that a wider multimedia streaming service can be supported, and there are now many ways in which TV channels can communicate with different terminals. Generally, scalable video streaming is known to provide more efficient channel capacity than simulcast video streaming. Simulcast video streaming requires a large network bandwidth for all resolutions, but scalable video streaming needs only one flow for all resolutions. In previous research, scalable video streaming has been compared with simulcast video streaming for network channel capacity, in two user simulation environments. The simulation results show that the channel capacity of SVC is 16% to 20% smaller than AVC, but scalable video streaming is not efficient because of the limit of the present network framework. In this paper, we propose a new network framework with an SVC extractor. The proposed network framework shows a channel capacity 50% (maximum) lower than that found in previous research studies.     

Cable-TV-based home video streaming system: practice and experience

This article presents a practical home video streaming system that has actually been implemented and deployed for commercial services on a cable television network. Because streaming components and network connections of a cable TV network are considerably different from those of desktop streaming systems available on the Internet, traditional systems and techniques for video streaming are not enough for a cable TV network. Therefore, several special systems and server-centric streaming techniques have been developed for a cable TV network and home video clients. In particular, server-centric streaming techniques are crucial because a set-top box, which is the client in the case of a home video streaming system, lacks processing power. We have developed a highly efficient streaming server using network acceleration and data bypassing techniques. As a result, our stateof- the-art streaming server enables high scalability and provides various content navigation modes, including faster playback, to off-the-shelf set-top boxes.     

End-to-end TCP-friendly streaming protocol and bit allocation for scalable video over wireless Internet

With the convergence of wired-line Internet and mobile wireless networks, as well as the tremendous demand on video applications in mobile wireless Internet, it is essential to an design effective video streaming protocol and resource allocation scheme for video delivery over wireless Internet. Taking both network conditions in the Internet and wireless networks into account, in this paper, we first propose an end-to-end transmission control protocol (TCP)-friendly multimedia streaming protocol for wireless Internet, namely WMSTFP, where only the last hop is wireless. WMSTFP can effectively differentiate erroneous packet losses from congestive losses and filter out the abnormal round-trip time values caused by the highly varying wireless environment. As a result, WMSTFP can achieve higher throughput in wireless Internet and can perform rate adjustment in a smooth and TCP-friendly manner. Based upon WMSTFP, we then propose a novel loss pattern differentiated bit allocation scheme, while applying unequal loss protection for scalable video streaming over wireless Internet. Specifically, a rate-distortion-based bit allocation scheme which considers both the wired and the wireless network status is proposed to minimize the expected end-to-end distortion. The global optimal solution for the bit allocation scheme is obtained by a local search algorithm taking the characteristics of the progressive fine granularity scalable video into account. Analytical and simulation results demonstrate the effectiveness of our proposed schemes.     

浅谈IPTV技术

IPTV是一种最新流行的、以“电视机+机顶盒”为主要终端设备、能为用户提供互动多媒体服务的宽带增值业务,它为我国广大电视用户提供了全新的宽带体验方式。IPTV技术是一项系统技术,包括多方面的内容,文章主要介绍了IPTV技术中的音视频编解码、流媒体传送、宽带接入网络、IP机顶盒等技术。     

网络流媒体传输中的自适应技术

如何在现有网络条件下进行多媒体音视频数据的有效传输控制是目前网络技术发展的一个热点。现在较多地采用流媒体技术来解决网络多媒体传输和播放的实时性需求，而自适应流媒体技术则是在不稳定的网络条件下进行数据传输有效控制的一种新技术。     

Network-centric music performance: practice and experiments

Advances in information technology and the great proliferation of the Internet have changed nearly every aspect of the work and life of human beings. Despite progress in networked entertainment, many music professionals and enthusiasts are still sticking to the traditional way of carrying out rehearsals and concerts. Music performance in this way requires physical presence of the participants and has a number of inherent limitations. We introduce a novel system called network-centric music performance (NMP) that enables multiparty music performance through cyberspace. Our target is to support real-time multichannel natural audio streaming over the network, using audio compression schemes that can provide acceptable audio quality. A system like this is bandwidth-demanding and highly delay-sensitive, and requires synchronization of the audio streams. Hence, support from the underlying end systems and networks is critical. However, the current source coding mechanisms and the best effort nature of the internet pose many challenges to achieve the desired quality of service. We have implemented a prototype of NMP, and exploited end system and network influences on NMP. The work was done in a LAN environment using Linux PCs. The system enables two different application scenarios: real-time rehearsal and rehearsal on demand. Real-time multichannel audio transport and different audio compression schemes are supported. Our evaluation results based on both subjective and objective measurements show that the system provides sufficient audio quality level for the target application in such an environment. The scalability test also revealed that the system scales well with increase of clientele. In the future, we extend our system for networks spanning larger distances and experiment with more realistic network conditions in the Internet.     

ViewStation applications: implications for network traffic

This paper describes applications built on the ViewStation, a distributed multimedia system based on Unix workstations and a gigabit per second local area network. A key tenet of the ViewStation project is the delivery of media data not just to the desktop but all the way to the application program. As processing power continues to improve, our approach enables applications that perform intensive processing of audio and video data. We hypothesize that as media data are shaped by this software-based processing, the resultant network traffic patterns will be dominated more by software behavior than by so-called real-time issues. We have written applications that directly process live video to provide more responsive human-computer interaction. We have also developed applications to explore the potential of media processing to support content-based retrieval of prerecorded television broadcasts. These applications perform intelligent processing on video, as well as straightforward presentation. They demonstrate the utility of network-based multimedia systems that deliver audio and video data all the way to the application. The network requirements of the applications are modeled as a combination of bursty transfers and periodic packet-trains     

A buffer‐aware fleet‐based cooperative H.264/SVC streaming over vehicular networks

Because of of the characteristics of high mobility, time varying and dynamic topology, how to provide multimedia streaming service for connected vehicles becomes one emerging and popular technical research. The motivation of this paper is to utilize cooperation among neighboring vehicles for video streaming's quality improvement over vehicular networks. In the proposed cooperative streaming scenario, a connected vehicle requests a video stream from the Internet by using its 3G/3.5G interface, which may not have enough bandwidth to receive good quality of video. Thus, the vehicle is suggested to ask neighboring members belonging to the same fleet to download the requested video data by using their 3G/3.5G interfaces. Then, neighboring members should forward video data to the requested vehicle by using another wireless technique, for example, dedicated short range communication (DSRC). Regarding the differentiation between the two access networks, that is, 3G/3.5G network and DSRC network, a buffer‐aware scheduling mechanism based on layered streaming is designed in this paper to adapt to the networking situation of the vehicular networks. Two selection algorithms are proposed to select neighboring vehicles from the fleet. According to our simulation results, the 3G/3.5G‐based selection algorithm is suitable to improve video quality for vehicles at low speeds. On the other hand, the DSRC‐based selection algorithm can get better performance when vehicles move at high speeds or too many data are transmitted among vehicles.Copyright © 2013 John Wiley & Sons, Ltd.     

基于TMS320DM642的网络视频监控服务器的研究与实现

基于TMS320DM642DSP处理器构建的硬件平台,采用JPEG图像压缩标准,系统在软件设计上采用了TI的RF5框架和DSP／BIOS实时内核,采用了扩展的“类驱动／微驱动”二级设备驱动程序模型,实现了一个具有实时视频监控和以太网传输功能的网络视频监控服务器。